Electrically tunable crystal band pass filters

[rhb]

It's common practice to use a capacitor to vary the frequency of a crystal over a narrow range in a VCXO or to trim the frequency in a Clapp or Colpitts design.

The same physics applies applies to  using a crystal as a filter, but I've not seen mention  of tunable crystal IF filters.

Consider a 5 MHz IF.  The spec on ebay 10-15 cent HC-49S crystals is +/- 50 ppm Fc.  That works out to 250 Hz which is irrelevant if you are moving the frequency by 1 kHz

I've attached the section of Zverer's "Handbook of Filter Synthesis".  The last page shows the filter I want to build as a tunable center frequency, constant BW filter which I can cascade with a similar filter with the opposite shift using an emitter follower to isolate the 2 filters.

It seems to me that if the caps in parallel with the xtals are implemented with varactors one can electrically tune the filter center frequency.  The BW needs to be 1500 Hz  to get good time domain response and ideally symmetric pass bands or anti-symmetric so that the intersection pass band is symmetric.

The Icom 705 does this in DSP, but this should be a lower power drain method of implementing the feature.

Does anyone have any experience with this concept or know of prior art?

Have  Fun!
reg

[David Hess]

More commonly a variable passband crystal filter is implemented with one or two fixed crystal filters, and varying the frequency of the local oscillator to the mixers.

[rhb]

Would you cite some examples?  I'm having a hard time grasping how it works.

Edit:  How would this work for a NorCal 40 style radio?

[David Hess]

It is called variable passband tuning.

For one IF filter, if the local oscillator for the first mixer is shifted, and the local oscillator for the second mixer is also shifted, then it is as if the IF filter is moving in frequency.

If this is done with two IF filters, then the combined width can also be changed because one filter will cutoff the low frequency, and the other will cutoff the high frequencies, producing a filter response narrower than either filter alone.

[T3sl4co1l]

Would it not be better to construct a standard crystal resonator filter and vary the coupling factor (thus bandwidth) instead?  That's simply varying the shunt capacitances.

Or are you looking to make two independently tunable filters, which might be used in cascade but could also be independent therefore should be independently tunable?

Tim

[rhb]

Please forgive my being especially dense today, but I'm  still baffled.

How does varying the frequency vary the width of the pass band?  It will shift the frequency of the input signal within the pass band, but doesn't change the combined pass band of the two filters.  So white noise input wouldn't change however you shifted the LO. 

The only way I can see what you describe  working is with another IF.  With which you get more IMD products.

The Icom 705 uses DSP to allow CW filter BWs from 50 to 1500 Hz in 50Hz steps.  And I've experienced complete rejection of a stronger signal  30 Hz away during a contest.  Icom calls it "Twin PBT".  You can hear the noise level drop as you narrow the filter overlap.  It's a cascade of a pair of 1500 Hz filters BW which can be shifted +/- 750 Hz.  The SSB filters are different.

It blew my mind.  I've been doing DSP for 40 years and had never heard of the idea.  You can have narrow filters but they ring badly.  So we never used them.  We never thought of running two broad filters with variable overlap. I'm still amazed. 

My plan is to code up Zverer's calculation for the 2 xtal filter and then solve for the same crystal at multiple center frequencies.  Then use that to develop a design to split a batch of xtals into high and low and then combine them to achieve the objectives using matched pairs.  At least until I figure out the mathematics of different xtal properties.

Have Fun!
Reg

[rhb]

Would it not be better to construct a standard crystal resonator filter and vary the coupling factor (thus bandwidth) instead?  That's simply varying the shunt capacitances.

Or are you looking to make two independently tunable filters, which might be used in cascade but could also be independent therefore should be independently tunable?

Tim

You can have narrow bandwidth or short time domain response, but not both from a single filter.

The time domain response of a BP filter is dominated by the BW.  But a broad band filter presented with a pure tone is going to have the time domain response matching its BW. 

I've attached the page from the Icom manual.

Reg

[T3sl4co1l]

You can have narrow bandwidth or short time domain response, but not both from a single filter.

The time domain response of a BP filter is dominated by the BW.  But a broad band filter presented with a pure tone is going to have the time domain response matching its BW. 

I've attached the page from the Icom manual.

Reg

Implying, what, that two filters together can violate Heisenberg* conditions?  If it's looks and smells like a free lunch, I got news for 'ya...

*The uncertainty principle in QM is just the time-frequency tradeoff mentioned above.  I... gosh, I forget offhand if there's a word/name/phrase associated with the pure Fourier version? But its physics application is well known, in any case.

If you capture the signals from such a receiver, or synthesize the filter yourself through whatever means, decomposable however you like -- you will find exactly such a duration of transient response.  The easiest way to see this, I suppose, is that the time-domain response of both filters is convolved, so the FIR kernel duration is the sum of the two for starters; and if each filter is sharp enough (say, high order Chebyshev) to give a usefully narrow bandwidth when so combined, then their transient responses will reflect that, being similarly long -- some times the reciprocal bandwidth.  In fact, that scale factor will be the Q factor of the highest-Q pole pair, or something similar to that.

Tim

[mtwieg]

TSP has a good video explaining different tunable filter types, including frequency-translating filters (that part starts at 16:50):

[David Hess]

Please forgive my being especially dense today, but I'm  still baffled.

How does varying the frequency vary the width of the pass band?  It will shift the frequency of the input signal within the pass band, but doesn't change the combined pass band of the two filters.  So white noise input wouldn't change however you shifted the LO.

Each filter is preceded by its own mixer and tunable local oscillator.  The combined response of the two filters is their overlap.  So if one filter is shifted lower in frequency, and the other filter is shifted higher in frequency, the combined passband becomes narrower.

This was a high end feature of triple conversion receivers which required precision tuning of each local oscillator, which became feasible with DDS.

[rhb]

@Tim  So how did I get complete rejection of a stronger CW signal 30 Hz offset from the station I was listening to without a trace of ringing?  DSP is not magic.  I could open the BW to 100 Hz and hear both or reduce it to 50 and here only one station. No change in time domain response from the full 1500 Hz to 50 Hz

@Dave what multi-conversion radio provided a 50 Hz CW BW filter option?

Most things which can be done using DSP can be done by analog means.  This is my best guess at present as to how to implement an analog equivalent.  If no one is interested in xtal filters which can be tuned electrically over a narrow range it's probably best I wander off.

Have  Fun!
Reg

[mawyatt]

Most things which can be done using DSP can be done by analog means.  This is my best guess at present as to how to implement an analog equivalent.  If no one is interested in xtal filters which can be tuned electrically over a narrow range it's probably best I wander off.

Have  Fun!
Reg

The fundamental difference between analog and digital filters, is analog filters are constrained as IIR types and digital filters can be FIR types.

Best,

[David Hess]

@Dave what multi-conversion radio provided a 50 Hz CW BW filter option?

Was that directed to me?  The narrowest filter option I commonly see is 250 Hz.  With passband tuning, it could potentially be narrower, but probably not with slopes as good as a dedicated narrow filter.

Most things which can be done using DSP can be done by analog means.  This is my best guess at present as to how to implement an analog equivalent.  If no one is interested in xtal filters which can be tuned electrically over a narrow range it's probably best I wander off.

There are still some differences.  While a practically ideal filter can be implemented using DSP, analog filtering has the advantage of removing interfering signals before they are further amplified and potentially cause overload, so dynamic range is higher.

[mawyatt]

A analog filter (IIR) is constrained such that if one wishes a maximally flat Amplitude response, then the Phase/Group Delay is already defined (Butterworth Type), or if a maximally flat Phase/Group Delay is required, then the Amplitude response is defined (Bessel). All the analog filters reside between these two extremes.

Whereas in the FIR digital filter domain the Amplitude and Phase/Group Delay can be independent. It's also usually much better to design a FIR digital filter for a given task rather than implement a classic analog type in the digital domain because of the analog limitations.

Best,

[mawyatt]

There are still some differences.  While a practically ideal filter can be implemented using DSP, analog filtering has the advantage of removing interfering signals before they are further amplified and potentially cause overload, so dynamic range is higher.

Good point about the pre analog filtering before the ADC conversion. We often implemented a typical front end analog filter with intent of limiting out-of-band signals sufficiently to not cause issues with the ADC conversion, then the high performance channel, signal, waveform selecting and such filtering in the digital domain, sometimes even "equalizing" the front end analog filter characteristics.

Best,

[rhb]

This thread is about tuning the two crystal band pass filter design in the attachment to my first post.

Got that?

My plan is to write a program to implement the design procedure given by Zverer  Then using the same crystal parameters solve for the capacitors required for the same BW and shifted center frequencies.  Then use a varactor to implement the values.

Anyone with something to say about the design process given in Zverer I'd love to hear from you.  The rest can have fun together.  I'm not responding further to anything outside the scope stated in bold.

Reg

[mawyatt]

Zverev is indeed the classic filter book for serious analog filter work and has a large section on just Crystal Filters alone from pg 414 to 498, and a host of different design approaches and procedures to consider.

The method you've shown in Fig 8.2 seems reasonable for the narrow band approach @ 5MHz, and maybe note the "Limitations on the Method Based..." regarding the transformer & other effects pg 452 and the note pg 449 about the shunt mutual transformer inductance not being considered.

Disclaimer we've not designed this type of narrowband crystal filter, so take our comments considering such, but "feel" the biggest obstacles ahead will be accurately characterizing the crystals eqv circuit models, not sure how well the cheap HC-49S are in this regard and require very high Q (see pg 452), and the actual transformers and their env models. As I'm sure you are aware small capacitance variations make a big difference in the filter overall response, especially so for a high "Q" narrowband type, so employing "trimmer" types to "home in" on the desired performance might be wise before introducing the varactors.

A simple terminated Emitter Follower (2N3904) should allow acceptable isolation between filter stages, altho a BUF34 or unity gain HS op-amp might prove better.

Anyway, good luck with your filter design!!

Best,

[T3sl4co1l]

@Tim  So how did I get complete rejection of a stronger CW signal 30 Hz offset from the station I was listening to without a trace of ringing?  DSP is not magic.

Beats me.  30Hz notch sounds pretty ringy to me.  I'm not up on the perceptual features of notch filters, if for example a flatter (Bessel?) but higher order type (= even longer transient response!) is more innocuous.  You would certainly see much ringing on a scope with an impulse response, but you might not hear it that way because it's the lack of frequencies, not the presence thereof.  And if it's surrounded by noise or tones rather than impulses, you might not notice it at all -- it's a lack of noise or tones.

Not sure if you're just not aware -- ears are notoriously poor oscilloscopes, and not even that good of signal analyzers, due to myriad perceptual biases.  Ears are the last thing you should trust; measure always!

Tim

[rhb]

@mawyatt Thank you for some excellent comments.  My fastest op amps are TL082s at 3 MHz.  Probably should do something about that ;-)  I'll use a 2N3904 for now.

Zverev presumes identical crystals.  I'd really like to simply adjust other circuit values to meet filter requirements.

In the presumption that if the crystals are not identical I *think* I can treat it as a coupled LC filter.  Thus with the  presence of a low power MCU (e.g. MSP430) to control the varactors what I thought to do was assemble parts,  derive the motional parameters of the xtals by sweeping the filter at several varactor settings.  Then explicitly calculate the required capacitance in parallel with the filter for a range of frequencies (2 kHz max). Then program the MCU to set a static DAC to produce that voltage when you set that frequency.  Add fixed and variable caps as needed based on varactor range. 

The nanoVNA should handle all the data collection and if you're selecting the best 4 from a batch of 10 it seems attractive to me.  But use L0 optimization to determine the optimal values using a PC program.

What I'm after is a "Not a NorCal 40" with twin pass band tuning in the analog domain at under 200 mA @ 12 V in receive.  That's 30% of the 705 drain.

BTW I gave 500 5 MHz xtals coming.

[rhb]

@Tim  So how did I get complete rejection of a stronger CW signal 30 Hz offset from the station I was listening to without a trace of ringing?  DSP is not magic.

Beats me.  30Hz notch sounds pretty ringy to me.  I'm not up on the perceptual features of notch filters, if for example a flatter (Bessel?) but higher order type (= even longer transient response!) is more innocuous.  You would certainly see much ringing on a scope with an impulse response, but you might not hear it that way because it's the lack of frequencies, not the presence thereof.  And if it's surrounded by noise or tones rather than impulses, you might not notice it at all -- it's a lack of noise or tones.

Not sure if you're just not aware -- ears are notoriously poor oscilloscopes, and not even that good of signal analyzers, due to myriad perceptual biases.  Ears are the last thing you should trust; measure always!

Tim

50 Hz BP centered on signal  with adjacent signal offset 5 Hz > than edge of filter. BTW I'm watching all this on the waterfall and changing the BW.  No notch filter involved.  Read the brochure page it's an accurate description.

I'd have thought I'd said this enough.  I was introduced to DSP in 1982 when I took a job as a geophysicist with Amoco.  I'd never heard of it or seen a seismic section. I then spent 4 years working under Milo Backus, one of Norbert Wiener's GAG students at UT Austin.  I then worked for most of the rest of my career as a contract scientist/programmer for big oil companies including super majors.  I attended 25 of 28 annual meetings, presented one paper and generally hung out with the A list in reflection seismology.

What I observed during the contest the weekend after I got the Icom 705 blew my mind.  You can't make 50 Hz BW filters!!!!!!!!!  But there I was confronted by the unarguable evidence.  So I had to figure out how this was possible.  If someone with my background goes, "WOW!!!" you might want to pay attention.  I immediately understood how it worked and why it was possible.  And astounded that I'd never heard of the technique before.

The time and frequency domain characteristics of a filter  are bound together by the Fourier transform and the minimum phase constraint of physics.

Reg

[mawyatt]

BTW Zverev indicates that if the bandwidth is 1% or lower, then the crystal "Q" becomes the limiting factor and things don't follow the design equations well, think was on pg 452. Your bandwidth is more like 0.03%, which might suggest that you can't achieve the desired narrow band results because of the limiting crystal "Q".

With 500 crystals you'll have a good range to select from, hopefully some very high "Q" ones in the batch. Do you have access to a quality LCR meter to extract the crystal parameters? I think a VNA might not give as good a result since the crystal parameters are in the high impedance range and far from the normalized 50 ohms, but maybe you can get a decent estimate of such with a VNA.

Regarding the emitter follower, run the 2N3904 at a high current so the emitter impedance is very low, then create an effective 50 ohm source with a ~50 ohm series R which should look reasonable symmetrical with sink/source dynamic currents as "seen" from the load side. You want the reflected energy from the 2nd filter stage input to "see" a 50 real source impedance for both + and - signal swings and absorb that energy, otherwise the bandpass and stopband will get distorted.

Anyway, we'll offer to evaluate a few crystals (not all 500 tho!!) with our Hioki IM3536 Lab LCR meter for you if you want, but shipping might be expensive.

Best,

[mawyatt]

What I observed during the contest the weekend after I got the Icom 705 blew my mind.  You can't make 50 Hz BW filters!!!!!!!!!  But there I was confronted by the unarguable evidence.  So I had to figure out how this was possible.  If someone with my background goes, "WOW!!!" you might want to pay attention.  I immediately understood how it worked and why it was possible.  And astounded that I'd never heard of the technique before.

The time and frequency domain characteristics of a filter  are bound together by the Fourier transform and the minimum phase constraint of physics.

Reg

Have you considered a Commutating Filter? We utilized this concept back in ~1982 to extract narrowband signals, where narrowband as in 1ppm! Bandwidth and Center Frequency completely tunable over decades of frequency and spot on because it's clock derived!

This is an interesting concept that straddles analog and digital domains with discrete time continuous amplitude signal processing.

Here's a note from memory, you can make the filter bilateral (add another series R on output side of C), the center frequency tracks the clock, the switches are just standard CMOS altho one could use (we did) much faster switches. You can experiment with just using Rin and Rout without the series R, and also you can make a bandreject (notch) by swapping R for series C altho we never did the notch type, maybe a good candidate for an experiment 

This Commutating Filter concept lay dormant for decades until it was realized that cutting the filter in half, and weighted summing the stored charge across each shunt capacitor created a new type RF/MW mixer that violated conventional physics and produced biphase passive mixers that had well below theoretical noise figures, we called these "Polyphase Mixer". Here's some notes on such, be sure to follow all the IEEE papers especially Andrews and Molnar who were the principle academic researchers at Cornell (our efforts like most weren't published due to the nature of our work), when this got out it lit up the entire academic research world overnight

Caution for those interested this is a deep technical dive, and one must spend considerable time and effort to truly understand and appreciate what's going on with this apparently simple circuit following the discrete time continuous amplitude path. Please follow all the IEEE papers, and discussions. So interesting and intriguing was this concept that DARPA created a special workshop on this very subject

https://www.eevblog.com/forum/rf-microwave/polyphase-or-n-path-mixer/msg3381802/#msg3381802

Best,

[rhb]

Ooh!  You're fun!  Lots of reading to do.  Thanks!

I'm fairly familiar (i.e. read an MS thesis) with commutating mixers, but never heard of commutating filters.  Same or different?  Who developed  this?  I can see lots of opportunity for a commutating mixer being a filter in the right hands.  A commutating mixer followed by a BP filter is a tunable BP filter.  Is that what you are describing?

The overall concept I'm pursuing is a 5 MHz "Twin Pass Band Tuning" IF followed by a quadrature detector (e.g. Tayloe)  with possibly additional filtering at audio using SCAFs or DSP on low power MCUs if a pair of 2 xtal filters don't provide adequate rejection. 

Edit: I neglected to look at the jpg.  Yes, that is exactly what I have in mind for the 2nd mixer.  And what I would describe as a commutating mixer followed by a BP filter.  I want to have analog with the option of DSP

[mawyatt]

The Polyphase or as some prefer the N-Path Mixer is not a traditional commutating mixer as you've studied.

First off the PPM achieves well below the theoretical 3.92dB NF of a biphase mixer as we were taught and as Fourier Analysis predicts (can explain if interested, and there's no LNA, this is completely passively), has very high TOI, bidirectional, complex (R +-jX) impedance matches at the input antenna port with baseband components and no inductors, input match tracks clock over decades of frequency without tunable elements (no varactors and such), directly down converts to I and Q at baseband and so on!! There's probably 50 or more research papers related to the PPM and it's derivatives now.

If you read and follow the mentioned Polyphase Mixer threads and details you can forgo the Tayloe Detector and Commutating Mixer all together, and just use a PPM. You can even create the narrow bandpass (1500Hz at 5MHz, and tune to 1MHz or 10MHz with constant bandwidth) at the Antenna Input as seen from RF side so out of band signals get squashed right at the antenna port!!

The research work dates back almost couple decades now but has trickled down to some of the higher end SDRs. It's complelely integratabtle in advanced CMOS, heck we were using TSMC 65nm CMOS decades ago for early renditions. Some time ago Apple acquired a small company (Dr Caroline Andrews was a principle) that was developing these devices and believe the Apple EarPods now utilize the PPM, likely in the iPhones as well. 

Anyway, fun stuff indeed

Best,

[mawyatt]

I'm fairly familiar (i.e. read an MS thesis) with commutating mixers, but never heard of commutating filters.  Same or different?  Who developed  this?  I can see lots of opportunity for a commutating mixer being a filter in the right hands.  A commutating mixer followed by a BP filter is a tunable BP filter.  Is that what you are describing?

Recall the 1st papers on the Commutating Filter were somewhere back in 1950~60s, they used a pair of automotive distributors (for V8) and drove the rotors on a common shaft with a synchronous motor.

BTW Biphase mixing, ie. multiplying by +-1, is the most efficient means of frequency mixing as taught and Fourier Analysis shows, and the translated frequency has a 2/pi amplitude loss, thus the theoretical 3.92dB Noise Figure. Since you can't get an ideal lossless conventional passive mixer, most high performance passive mixers when driven by a high power LO (approximating a +-1 multiplication) show NF of ~6dB. Active mixers (i.e. Gilbert style) are much worse since the also modulate a DC bias which introduces additional noise and have NF greater than ~9dB. This was all true until the discovery of the PPM which completely displaces this mixer theoretical NF limit, and believe some have achieved (measured) ~1dB NF to date.

The physics behind this PPM that can trump mixer noise theory is fascinating and difficult to get ones mind wrapped around, but nevertheless does so wether one understands such or not, the PPM knows

The overall concept I'm pursuing is a 5 MHz "Twin Pass Band Tuning" IF followed by a quadrature detector (e.g. Tayloe)  with possibly additional filtering at audio using SCAFs or DSP on low power MCUs if a pair of 2 xtal filters don't provide adequate rejection.

You can do all this and much more with a single PPM, and likely even eliminate the entire RF front end as the PPM is a Direct Down Conversion Mixer with Direct I and Q outputs!!

Edit: I neglected to look at the jpg.  Yes, that is exactly what I have in mind for the 2nd mixer.  And what I would describe as a commutating mixer followed by a BP filter.  I want to have analog with the option of DSP

The PPM is a case, as is the Commutating Filter, of what we coined back in 70s as Discrete Time Continuous Amplitude Signal Processing utilizing the benefits of both analog and digital signal processing. We developed a custom handheld Real Time SA back in ~80 utilizing this signal processing based upon the Chirp Z Transform with custom CCD chips as the real time convolvers operating with discrete time (clocked) continuous amplitude (charge domain).

Anyway, fascinating subject

Best,